Commit graph

23 commits

Author SHA1 Message Date
Alessio Bazzica
cc22f51988 Removing the intelligibility enhancer.
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).

Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
2018-08-30 21:29:57 +00:00
Minyue Li
656d609a95 Add UTC time to init event in AEC debug dump.
Bug: webrtc:9616
Change-Id: I1350212f0b8835fb64427483269da96d51670c01
Reviewed-on: https://webrtc-review.googlesource.com/92620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-11 20:29:07 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Alex Loiko
1aec594018 Merge :audio_processing and :aec_dump_interface.
Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502

':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.

Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.

NOTRY=true # want this in and 'ios_api_framework' seems stuck.

Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
2018-05-15 14:22:52 +00:00
Alex Loiko
5feb30e85f Options and settings for the Pre-amplifier.
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.

Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.

Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
2018-04-16 12:25:48 +00:00
Fredrik Solenberg
bbf21a3fd6 Remove dependencies on modules:module_api from AudioProcessing.
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.

Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:05:27 +00:00
Karl Wiberg
6a4d411023 Move file_wrapper.h to rtc_base/system/
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.

BUG=webrtc:8445

Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 11:17:15 +00:00
Patrik Höglund
7696bef463 Remove the public_deps to fileutils from test_support.
Bug: webrtc:8946
Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90
No-Presubmit: true
Reviewed-on: https://webrtc-review.googlesource.com/62100
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-16 09:06:27 +00:00
Alex Loiko
e36e8bbf6d Add FixedGainController and move GainController2 in APM.
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.

The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().

This CL contains

* build changes to make modules/audio_processing/agc2 an independent
  target

* a new MutableFloatAudioFrame which is the audio interface between
  AGC2 and APM

* move of the fixed gain application from GainController2 to
  FixedGainController.

If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#

Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 10:56:38 +00:00
Mirko Bonadei
ca913b0549 Stop using public_deps in modules/audio_processing/aec_dump.
Bug: webrtc:8603
Change-Id: I8d21a195323bfa088003d47a67f41a387d0101fa
Reviewed-on: https://webrtc-review.googlesource.com/34186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21791}
2018-01-29 13:13:08 +00:00
Mirko Bonadei
65ce31158f Removing useless dependencies on //testing/gmock.
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.

Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).

This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.

Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.

TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
2018-01-26 13:34:12 +00:00
Per Kjellander
a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
This reverts commit c73e1f4378.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00
Per Kjellander
c73e1f4378 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
This reverts commit 588c548657.

Reason for revert: 

Breaks Chrome FYI:

/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
  -> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
    static_library(target_name) {
    ^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
  //third_party/webrtc/*
  //third_party/webrtc_overrides/*
]

 https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout

Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> 
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
> 
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
> 
> BUG=webrtc:8254
> 
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
2018-01-10 15:14:54 +00:00
Karl Wiberg
588c548657 GN rtc_* templates: Set default visibility to webrtc_root + "/*"
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.

API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.

BUG=webrtc:8254

Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
2018-01-10 13:08:11 +00:00
Ivo Creusen
62337e59dd Use AudioProcessingBuilder everywhere AudioProcessing is created.
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.

Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
2018-01-09 13:45:20 +00:00
Patrik Höglund
a8005cfd8b Fix circular dependencies between optional, array_view, and rtc_base.
This splits things out of rtc_base and makes dependencies explicit.

Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-14 06:49:11 +00:00
Patrik Höglund
d37709b659 Revert "Fix circular dependencies between optional, array_view, and rtc_base."
This reverts commit a9e0924fa7.

Reason for revert: Breaks because of RTC_LAST_SYSTEM_ERROR

Original change's description:
> Fix circular dependencies between optional, array_view, and rtc_base.
> 
> This splits things out of rtc_base and makes dependencies explicit.
> 
> Bug: webrtc:6828
> Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
> Reviewed-on: https://webrtc-review.googlesource.com/31940
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21245}

TBR=phoglund@webrtc.org,kwiberg@webrtc.org

Change-Id: I1a5dcf2223f00ae7c46f9f2a12b990ab3a84397d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/32760
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21251}
2017-12-13 14:56:33 +00:00
Patrik Höglund
a9e0924fa7 Fix circular dependencies between optional, array_view, and rtc_base.
This splits things out of rtc_base and makes dependencies explicit.

Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
2017-12-13 13:44:21 +00:00
Mirko Bonadei
5b86f0a24b Stop using ByteSize (deprecated) to get the size of a proto message.
The method ByteSize has been deprecated [1], this CL switches to
ByteSizeLong.

[1] - https://cs.chromium.org/chromium/src/third_party/protobuf/src/google/protobuf/message_lite.h?l=252&rcl=ac47edd22c481fcfe119769d6b7abf365abea8fa

Bug: None
Change-Id: I1ba622df52f47719a5beda6d230cb603a0163d43
Reviewed-on: https://webrtc-review.googlesource.com/27021
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20952}
2017-11-30 14:27:50 +00:00
Alex Loiko
34fa309129 Twitter-famous NOOP bug.
Between patch set 4 and patch set 5 in
https://codereview.webrtc.org/2865113002/, a line consisting of a
single 'std::move(task);' was added. The reason we will never know,
because the author will not tell. The superfluous line would have gone
unnoticed except for occasional raised eyebrows of casual code
readers.

The Visual Studio compiler now sees lines that have no effect. Which
was announced to the world in the tweet
https://twitter.com/StephanTLavavej/status/924011366943354880
achieving 27 likes and 6 retweets.

Bug: webrtc:8463
Change-Id: Iac49bc4153254b6cfe99f609da28eb4f43ff765e
Reviewed-on: https://webrtc-review.googlesource.com/21324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20616}
2017-11-09 11:38:12 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00